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https://github.com/oxen-io/session-android.git
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79 lines
2.8 KiB
C
79 lines
2.8 KiB
C
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
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#ifdef AGC_DEBUG
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#include <stdio.h>
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#endif
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/typedefs.h"
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// the 32 most significant bits of A(19) * B(26) >> 13
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#define AGC_MUL32(A, B) (((B)>>13)*(A) + ( ((0x00001FFF & (B))*(A)) >> 13 ))
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// C + the 32 most significant bits of A * B
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#define AGC_SCALEDIFF32(A, B, C) ((C) + ((B)>>16)*(A) + ( ((0x0000FFFF & (B))*(A)) >> 16 ))
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typedef struct
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{
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int32_t downState[8];
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int16_t HPstate;
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int16_t counter;
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int16_t logRatio; // log( P(active) / P(inactive) ) (Q10)
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int16_t meanLongTerm; // Q10
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int32_t varianceLongTerm; // Q8
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int16_t stdLongTerm; // Q10
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int16_t meanShortTerm; // Q10
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int32_t varianceShortTerm; // Q8
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int16_t stdShortTerm; // Q10
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} AgcVad_t; // total = 54 bytes
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typedef struct
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{
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int32_t capacitorSlow;
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int32_t capacitorFast;
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int32_t gain;
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int32_t gainTable[32];
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int16_t gatePrevious;
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int16_t agcMode;
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AgcVad_t vadNearend;
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AgcVad_t vadFarend;
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#ifdef AGC_DEBUG
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FILE* logFile;
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int frameCounter;
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#endif
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} DigitalAgc_t;
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int32_t WebRtcAgc_InitDigital(DigitalAgc_t *digitalAgcInst, int16_t agcMode);
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int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *digitalAgcInst,
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const int16_t *inNear, const int16_t *inNear_H,
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int16_t *out, int16_t *out_H, uint32_t FS,
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int16_t lowLevelSignal);
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int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc_t *digitalAgcInst,
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const int16_t *inFar,
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int16_t nrSamples);
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void WebRtcAgc_InitVad(AgcVad_t *vadInst);
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int16_t WebRtcAgc_ProcessVad(AgcVad_t *vadInst, // (i) VAD state
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const int16_t *in, // (i) Speech signal
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int16_t nrSamples); // (i) number of samples
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int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
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int16_t compressionGaindB, // Q0 (in dB)
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int16_t targetLevelDbfs,// Q0 (in dB)
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uint8_t limiterEnable,
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int16_t analogTarget);
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
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